diff --git a/main/xiaozhi-server/core/connection.py b/main/xiaozhi-server/core/connection.py
index 373c977a..790e2879 100644
--- a/main/xiaozhi-server/core/connection.py
+++ b/main/xiaozhi-server/core/connection.py
@@ -314,7 +314,7 @@ class ConnectionHandler:
                 break
 
             end_time = time.time()
-            self.logger.bind(tag=TAG).debug(f"大模型返回时间: {end_time - start_time} 秒, 生成token={content}")
+            # self.logger.bind(tag=TAG).debug(f"大模型返回时间: {end_time - start_time} 秒, 生成token={content}")
 
             # 合并当前全部文本并处理未分割部分
             full_text = "".join(response_message)
@@ -433,7 +433,7 @@ class ConnectionHandler:
                         break
 
                     end_time = time.time()
-                    self.logger.bind(tag=TAG).debug(f"大模型返回时间: {end_time - start_time} 秒, 生成token={content}")
+                    # self.logger.bind(tag=TAG).debug(f"大模型返回时间: {end_time - start_time} 秒, 生成token={content}")
 
                     # 处理文本分段和TTS逻辑
                     # 合并当前全部文本并处理未分割部分
diff --git a/main/xiaozhi-server/core/handle/receiveAudioHandle.py b/main/xiaozhi-server/core/handle/receiveAudioHandle.py
index f97e1e23..a082ada8 100644
--- a/main/xiaozhi-server/core/handle/receiveAudioHandle.py
+++ b/main/xiaozhi-server/core/handle/receiveAudioHandle.py
@@ -10,7 +10,7 @@ logger = setup_logging()
 
 async def handleAudioMessage(conn, audio):
     if not conn.asr_server_receive:
-        logger.bind(tag=TAG).debug(f"前期数据处理中，暂停接收")
+        # logger.bind(tag=TAG).debug(f"前期数据处理中，暂停接收")
         return
     if conn.client_listen_mode == "auto":
         have_voice = conn.vad.is_vad(conn, audio)
diff --git a/main/xiaozhi-server/core/handle/sendAudioHandle.py b/main/xiaozhi-server/core/handle/sendAudioHandle.py
index c0a664eb..9dfa5bd2 100644
--- a/main/xiaozhi-server/core/handle/sendAudioHandle.py
+++ b/main/xiaozhi-server/core/handle/sendAudioHandle.py
@@ -36,7 +36,7 @@ async def sendAudio(conn, audios):
     pre_buffer = min(3, len(audios))
     for i in range(pre_buffer):
         await conn.websocket.send(audios[i])
-        conn.logger.bind(tag=TAG).debug(f"预缓冲帧 {i}, 时间: {(time.perf_counter() - start_time) * 1000:.2f}ms")
+        # conn.logger.bind(tag=TAG).debug(f"预缓冲帧 {i}, 时间: {(time.perf_counter() - start_time) * 1000:.2f}ms")
 
     # 正常播放剩余帧
     for opus_packet in audios[pre_buffer:]:
@@ -51,7 +51,7 @@ async def sendAudio(conn, audios):
             await asyncio.sleep(delay)
 
         await conn.websocket.send(opus_packet)
-        conn.logger.bind(tag=TAG).debug(f"发送帧，位置: {play_position}ms, 实际间隔: {(time.perf_counter() - current_time) * 1000:.2f}ms")
+        # conn.logger.bind(tag=TAG).debug(f"发送帧，位置: {play_position}ms, 实际间隔: {(time.perf_counter() - current_time) * 1000:.2f}ms")
 
         play_position += frame_duration